Re: [asterisk-users] Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)
Max Alex wrote: > Hi Hiren, > Can you please confirm the php-gd is properly installed? If you can't export to pdf that's because of a typo in export_pdf.php in line 5: include_once(dirname(__FILE__) . "/lib/fpdf.php'); Note that it starts with " and ends with ' , so it gives an error. It works by changing it to this: include_once(dirname(__FILE__) . "/lib/fpdf.php"); Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie AEL2: Syntax for Hint
I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Help
Hi tecnocrat,
I 'm trying to setup a Dundi system like yours (one lookup server and 2
pbx servers). I searched on internet for some docs but found a lot of
stuff explaining only a part of the problem and no good example at all
(there's a Richardson doc in internet which can help to start).
I tried to do it myself so I generated the two keys (pri and pub) for
each server with their own hostname then I copied:
- .121 keys to the other two servers (.137 and .204)
- .137 keys to .121
- .204 keys to .121
Let me know how if it works.
Giorgio Incantalupo
technocrat voip wrote:
> Hello All,
>
> Iam trying to achive a simple load balancing with dundi.
>
> Here i have three asterisk boxes like below.
>
>
> *.*.*.121 which is the dundi server
>
> *.*.*.137 "A" Peer which has the 1000 phone registerd to it
>
> *.*.*.204 "B" Peer which has the 200 phone registered to it.
>
> The expected behavior of my setup is once i dial from 1000 phone it
> has to goto "B" peer using the ".121" dundi server.
>
> Iam getting confused with the public key / private key stuff here.
>
> Iam using the "astgenkey -n " command to generate them .
>
> Can any body help me by explaining , what keys i have to generate on
> each server and which keys i need to copy to which server .
>
> regards
>
>
>
> All the conf file stuff is is like below
>
> *.*.*.137
>
> iax.conf
>
> [priv]
> type=friend
> dbsecret=dundi/secret
> context=incomingdundi
>
>
> dundi.conf
>
> [mappings]
> priv =>
> sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
>
>
> [00:0F:E2:76:4B:33]
> model = symmetric
> host = *.*.*.21
> inkey = dundincomingseven
> outkey = dundiseven
> include = priv
> permit = priv
> qualify = yes
> order = primary
>
>
> *.*.*.204
>
> iax.conf
>
> [priv]
> type=friend
> dbsecret=dundi/secret
> context=incomingdundi
>
> dundi.conf
>
> [mappings]
> priv =>
> sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
> ;
>
> [00:0F:E2:76:4B:33]
> model = symmetric
> host = *.*.*.21
> inkey = dundiincomingfour
> outkey = dundifour
> include = priv
> permit = priv
> qualify = yes
> order = primary
>
> *.*.*.21
>
> iax.conf
> [priv]
> type=user
> context=local-custom
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
>
>
> dundi.conf
>
>
> [mappings]
> priv => dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
>
> [00:11:0A:34:29:57]
> model = symmetric
> host = *.*.*.137
> inkey = dundi
> outkey = dundi
> include = priv
> permit = priv
> qualify = yes
> order = primary
>
> [00:11:0A:34:29:43]
> model = symmetric
> host = *.*.*.204
> inkey = dundi
> outkey = dundi
> include = priv
> permit = priv
> qualify = yes
> order = primary
> .
>
>
>
>
>
>
>
> ___
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>
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_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FG&A srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172
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Re: [asterisk-users] Dundi Help
hi gior,
If i understand correctly your setup would be like below.
A is the dundi serer
B is one pbx
C is one pbx
B and C dundi.conf contain the entity detials of A.
Either for C or B we can place calls to the extensions registered on the
other server.
When C extension make call to B extension call go through A and reach the B.
If you can provide the details of dundi.conf and the extension.conf it would
be very help full.
to dig into the issue.
could you able to do the dundi lookup ..?
regards
On Wed, Sep 10, 2008 at 3:36 PM, Giorgio Incantalupo <
[EMAIL PROTECTED]> wrote:
> Hi tecnocrat,
>
> I 'm trying to setup a Dundi system like yours (one lookup server and 2
> pbx servers). I searched on internet for some docs but found a lot of
> stuff explaining only a part of the problem and no good example at all
> (there's a Richardson doc in internet which can help to start).
> I tried to do it myself so I generated the two keys (pri and pub) for
> each server with their own hostname then I copied:
> - .121 keys to the other two servers (.137 and .204)
> - .137 keys to .121
> - .204 keys to .121
>
> Let me know how if it works.
>
> Giorgio Incantalupo
>
>
> technocrat voip wrote:
> > Hello All,
> >
> > Iam trying to achive a simple load balancing with dundi.
> >
> > Here i have three asterisk boxes like below.
> >
> >
> > *.*.*.121 which is the dundi server
> >
> > *.*.*.137 "A" Peer which has the 1000 phone registerd to it
> >
> > *.*.*.204 "B" Peer which has the 200 phone registered to it.
> >
> > The expected behavior of my setup is once i dial from 1000 phone it
> > has to goto "B" peer using the ".121" dundi server.
> >
> > Iam getting confused with the public key / private key stuff here.
> >
> > Iam using the "astgenkey -n " command to generate them .
> >
> > Can any body help me by explaining , what keys i have to generate on
> > each server and which keys i need to copy to which server .
> >
> > regards
> >
> >
> >
> > All the conf file stuff is is like below
> >
> > *.*.*.137
> >
> > iax.conf
> >
> > [priv]
> > type=friend
> > dbsecret=dundi/secret
> > context=incomingdundi
> >
> >
> > dundi.conf
> >
> > [mappings]
> > priv =>
> > sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
> >
> >
> > [00:0F:E2:76:4B:33]
> > model = symmetric
> > host = *.*.*.21
> > inkey = dundincomingseven
> > outkey = dundiseven
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> >
> >
> > *.*.*.204
> >
> > iax.conf
> >
> > [priv]
> > type=friend
> > dbsecret=dundi/secret
> > context=incomingdundi
> >
> > dundi.conf
> >
> > [mappings]
> > priv =>
> > sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
> > ;
> >
> > [00:0F:E2:76:4B:33]
> > model = symmetric
> > host = *.*.*.21
> > inkey = dundiincomingfour
> > outkey = dundifour
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> >
> > *.*.*.21
> >
> > iax.conf
> > [priv]
> > type=user
> > context=local-custom
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> >
> >
> > dundi.conf
> >
> >
> > [mappings]
> > priv => dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
> >
> > [00:11:0A:34:29:57]
> > model = symmetric
> > host = *.*.*.137
> > inkey = dundi
> > outkey = dundi
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> >
> > [00:11:0A:34:29:43]
> > model = symmetric
> > host = *.*.*.204
> > inkey = dundi
> > outkey = dundi
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> > .
> >
> >
> >
> >
> >
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
>
> _
> Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
> FG&A srl - http://www.fgasoftware.com -
> [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
> Tel: 02997663.14, Fax: 0291390172
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
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>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] How to make outgoing call from Pri ?
Dear Everyone, I have a one Outgoing pri. And I have to use this pri for outgoing call from this premises. Can anyone guide me how it is done and what I have to mention in extention.conf If I have to get zero dialing facility for out going. -- With Regards, Hiren Mistry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resilience using DNS or phone feature ?
Hi, I'm planning to deploy SIP hardphones in a serverless location. Phones would be connected to 2 different Asterisk servers, one backing up the other. I would like to offer resilience and I'm wondering about the best way to do it. Phones themselves can register to a backup SIP proxy if first proxy fails but, AFAIK, can't fall back to main server from backup server when main server recovers. I'm wondering if should use DNS, phone multi-registration feature, or a combination of DNS and phone multi-registration feature, to implement resilience. Your opinion ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Help
technocrat voip wrote:
> hi gior,
>
> If i understand correctly your setup would be like below.
>
> A is the dundi serer
> B is one pbx
> C is one pbx
yes
>
> B and C dundi.conf contain the entity detials of A.
yes
>
> Either for C or B we can place calls to the extensions registered on
> the other server.
tried only from B to C just for test. Sure it will work from C to B too.
>
> When C extension make call to B extension call go through A and reach
> the B.
In theory it should, but the A console shows nothing when calling...do
not know why...
>
> If you can provide the details of dundi.conf and the extension.conf it
> would be very help full.
> to dig into the issue.
Sure but it is a lot of stuff (I post it below...it is a bit long ::)
Hope may help.)
> could you able to do the dundi lookup ..?
The dundi lookup works fine (not locally, only from one machine to
others according to wiki), but I've got problems to understand some part
of the files (es: default and peer sections).
>
> regards
::)
---
- Server DundiLookup -
---
DUNDI.CONF
[general]
department=dundio
organization=dundio
locality=Rho
stateprov=MI
country=IT
[EMAIL PROTECTED]
phone=0123456789
entityid=00:30:84:7A:5B:FA
cachetime=5
ttl=2
autokill=yes
[mappings]
dundi-priv => sipregistration,0,SIP,[EMAIL PROTECTED],nopartial
;per chiamare via sip
[00:30:84:7A:55:9F]
model = symmetric
host = 192.168.0.51
inkey = 51
outkey = 71
include = dundi-priv
permit = dundi-priv
qualify = yes
dynamic= yes
order = primary
[00:30:84:41:0C:1C]
model = symmetric
host = 192.168.0.55
inkey = 55
outkey = 71
include = dundi-priv
permit = dundi-priv
qualify = yes
dynamic= yes
order = primary
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[default]
exten => _XXX,1,Macro(dundi-priv,${EXTEN})
exten => _XXX,2,Playback(invalid)
exten => _XXX,3,Hangup
[iax-clt]
include => sipregistration ;include i sip dell'extensions locale
exten => _XXX,2,Answer
exten => _XXX,3,Dial(SIP/${EXTEN})
exten => _XXX,4,Hangup
[dundi-priv-local]
include => iax-clt
[dundi-priv-switch]
switch => dundi/dundi-priv
[dundi-priv-lookup]
include => dundi-priv-local
include => dundi-priv-switch
[macro-dundi-priv]
exten=> s,1,goto(${ARG1},1)
include => dundi-priv-lookup
SIP.CONF
[general]
regcontext=sipregistration
[peer]
username=peer
context=default
context=default; Default context for incoming calls
bindport=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes; Enable DNS SRV lookups on outbound calls
---
- PBX server -
---
DUNDI.CONF
[general]
department=test
blah blah
entityid=00:30:84:7A:55:9F
cachetime=5
ttl=2
autokill=yes
[mappings]
dundi-priv => dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
;call via sip
[00:30:84:7A:5B:FA]
model = symmetric
host = 192.168.0.71
inkey = 71
outkey = 51
include = dundi-priv
permit = dundi-priv
qualify = yes
dynamic= yes
order = primary
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
autofallthrough=no
[default]
exten => _XXX,1,Macro(dundi-priv,${EXTEN})
exten => _XXX,2,Playback(invalid)
exten => _XXX,3,Hangup
[iax-clt]
include => sipregistration ;including local extensions
exten => _XXX,2,Answer
exten => _XXX,3,Dial(SIP/${EXTEN})
exten => _XXX,4,Hangup
[dundi-priv-local]
include => iax-clt
[dundi-priv-switch]
switch => dundi/dundi-priv
[dundi-priv-lookup]
include => dundi-priv-local
include => dundi-priv-switch
[macro-dundi-priv]
exten=> s,1,goto(${ARG1},1)
include => dundi-priv-lookup
[general]
regcontext=sipregistration ;mandatory
[peer] ; mandatory
username=peer
context=default
[100]
type = friend
secret=blah
context=default
host=dynamic
qualify = no
username=100
fromuser=100
dtmfmode = rfc2833
language = it
SIP.CONF
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Re: [asterisk-users] how to disallow the native bridge between the two channel
On Wed, Sep 10, 2008 at 9:48 AM, bala krishnan <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> Thanks for your apt response.
>
> I also tried by starting the recording before the dial command through AMI.
> But in that scenario also, there are some issue such that both the streams
> are not getting updated at the same time. Sometimes one stream takes nearly
> 2 seconds to getting updated. Here i require both the stream updation needs
> to be happening at the same time. And this voice updation issue is not there
> when i included the t option in Dial application.
No, what i mean - is to use dialplan command before you dial to
destination peer. For example:
Monitor(ulaw,/tmp/recording-${UNIQUEID},b);
Regards,
Atis
>
> Kindly give your suggestion on this.
>
> Asterisk version - 1.4.21.2
>
> Thanks,
> balasam.
> On Tue, 09 Sep 2008 Atis Lezdins wrote :
>>On Tue, Sep 9, 2008 at 3:19 PM, bala krishnan <[EMAIL PROTECTED]>
>> wrote:
>> > Hi,
>> >The problem is, when i was starting the recording on zap channels
>> > through AMI by Monitor command, always the out stream recorded as 0
>> > bytes.
>> > So that i did the searching and got the response that t option would
>> > disallow the native bridging between the channels and you will get the
>> > properly recorded out stream files. With that t option, now i am getting
>> > the
>> > properly recorded stream files.
>> >
>> > What i would want is, instead of t option in Dial application, is there
>> > any
>> > other way to set this in the configuration file?
>>
>>Hi,
>>
>>This should probably belong to list, as I don't have all answers :)
>>
>>As I understand, you don't want recording unless you start it by AMI.
>>
>>Theoretically AMI action Monitor should destroy native bridge and
>>re-bridge call trough Asterisk. It could be a bug, or just
>>unimplemented feature.
>>
>>So, i could suggest starting Monitor before Dial command into
>>temporary dir, and then erase unnecessary recordings.. That way, AMI
>>Monitor will simply override previous and you will get recording where
>>you want, and bridge won't be native.
>>
>>Regards,
>>Atis
>>
>> >
>> >
>> > regards,
>> > balasam
>> > On Mon, 08 Sep 2008 Atis Lezdins wrote :
>> >
>> >>On Mon, Sep 8, 2008 at 11:39 AM, bala krishnan <[EMAIL PROTECTED]>
>> >> wrote:
>> >> > Hi,
>> >> >
>> >> > To disallow the native bridge between the zap channels, i enabled
>> >> > the t
>> >> > flag in the Dial application. But i dont want to allow the
>> >> > callee/caller
>> >> > to
>> >> > transfer the call.
>> >>
>> >>Why would you need this? It should just take media processing away
>> >> from your CPU.
>> >>
>> >>Alternatively you can enable Monitor/MixMonitor, it should keep
>> >>Asterisk in media path.
>> >>
>> >>Regards,
>> >>Atis
>> >>
>> >>--
>> >>Atis Lezdins,
>> >>VoIP Project Manager / Developer,
>> >>[EMAIL PROTECTED]
>> >>Skype: atis.lezdins
>> >>Cell Phone: +371 28806004
>> >>Cell Phone: +1 800 7300689
>> >>Work phone: +1 800 7502835
>> >
>> >
>> >
>>
>>
>>
>>--
>>Atis Lezdins,
>>VoIP Project Manager / Developer,
>>[EMAIL PROTECTED]
>>Skype: atis.lezdins
>>Cell Phone: +371 28806004
>>Cell Phone: +1 800 7300689
>>Work phone: +1 800 7502835
>
>
>
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)
I use Amazon EC2 when my capacity reach the max limit. Because I don't have control on witch datacenter or Internet connection my new virtual machine will start I got some problems, not very often, with low bandwith and now I'm working on a new AMI with watchdogs for voice quality and latency issues. My plan is to use AWS just for transcoding... Maybe some other guys in community use it for IVRs or gateways... I'm curios to find out :) Best regards, Chris Hariga MyPBX. US --Original Message-- From: Steve Finkelstein Sender: To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sep 9, 2008 10:56 PM Subject: [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3) Hey folks, I'm looking to potentially take some of my Asterisk servers and see how well they fare in a cloud computing environment such as Amazon EC2 + S3. I was curious to hear feedback from anyone who's willing to share their experience if they've already done the same. Have you had a positive experience and if not with Amazon, what other grid computing platform? Was it horrible and you'll never go back to it? Great ordeal of jitter/noise? Thanks a lot for your insight. :-) /sf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent from my BlackBerry® wireless device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Network Monitoring
Zabbix- we even developed a small zabbis agent script to report back the status of PRIs, etc. Paul Hales wrote: > I have used both munin and nagios - both are cool. > > PaulH > > > > EdPimentl wrote: >> http://www.voip-info.org/wiki/view/Asterisk+monitoring >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue reload
On Tuesday 09 September 2008 12:30, Atis Lezdins wrote: > On Tue, Sep 9, 2008 at 1:22 PM, Thomas Winter <[EMAIL PROTECTED]> wrote: > > On Monday 08 September 2008 14:44, Atis Lezdins wrote: > >> On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter <[EMAIL PROTECTED]> > > > > wrote: > >> > I dont have problem to make a reload by AMI. > >> > My questions was if module reload app_queue.so is the right way to do > >> > this, because whis "reload" I reload everything. > >> > > >> > Its fact that I have to do reload queue otherwise Asterisk did not > >> > load realtime database with new settings. > >> > >> Definitely not. > >> > >> Realtime should reload settings on every new call, and this is working > >> for me on periodic_announce and periodic_announce_frequency. However > >> this will work only for new calls, existing calls will have settings > >> as loaded at their enter queue. > >> > >> My Asterisk version is 1.4.19, addons 1.4.6 > >> > >> If you enable debug 1 you should see in your full log: > > > > Hi, > > I have 1.4.21.2 and addons 1.4.7 > > > > If I do reload I have this: > > > > [Sep 9 12:10:12] DEBUG[4709] res_config_mysql.c: MySQL RealTime: Static > > SQL: SELECT category, var_name, var_val, cat_metr > > ic FROM fileconf WHERE filename='queues.conf' and commented=0 ORDER BY > > filename, cat_metric desc, var_metric asc, category > > , var_name, var_val, id > > > > And I have found only this in debug file: > > > > [Sep 9 12:12:02] DEBUG[20173] app_queue.c: Queue test has no realtime > > members defined. No need for update > > > > Might be this is the reason, I do add agents with AMI QueueAdd. > > > > So changes in realtime queues.conf will not be read, I have to do reload. > > Oooh, so you have "Static realtime". I think it isn't supposed to > reload automatically. Go for "Real Realtime" - > http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue or live > with "module reload" :) I see, I use module reload app_queue.so and it works fine. thanks best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote: > I am struggling to find out how to code hint in AEL2. > > I did hint(Custom:light1) and it keeps complaining about the : (colon). > It works fine for SIP device like hint(SIP/439). > > Anyone who has tried it before? Yes, a while back I upgraded AEL to handle both ':' and '&' inside the hint parens. This should work on 1.4 on up. What version of asterisk are you using? 1.2? murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue
Hi Folks, I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped. >From searching the list archives, I believe that a DMS wants to receive the CNAM info in the Q.931 call setup message. I've tried all permutations of switchtype (dms100 & national) and facilityenable that I can think of, but I still don't see CNAM coming out the other side. Telco confirms that "Name Out" is enabled on our PRI. Any pointers on what I'm missing, and/or how to debug further? zapata.conf: --- [channels] context=default callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no canpark=yes cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 ;facilityenable=yes immediate=no priindication=outofband switchtype=national context=from-pstn group=0 signalling=pri_cpe channel =>1-23 --- Regards, Iain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iLBC and G729 codecs
Edgar Guadamuz wrote: > I notice that I have only format_ilbc.so but not codec_ilbc.so... is > it due to the compilation or there is some way to create the module? > That's the format interpreter, for the codec you need to select it in make menuselect befor ecompiling asterisk (subject to libraries). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk phone conferencing performance
Yes, I am using the ztdummy and 1.6beta Asterisk. There is no load on the machine - hovers between 1%-2% CPU usage during the conference. Thank you. >You shouldn't have any delays at all. >Are you using ztdummy for timing? and what kind of load does the box >have on it? On Sep 9, 2008, at 4:23 PM, George Williams wrote: > Hi, > > I just set up my first Asterisk with MeetMe conference support on my > local LAN. > > It works great, but I think it may need a little tuning - I am > getting audio delays of up to 1 second. Should I expect better > performance in this area, or is this to be expected? > > Thanx! > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue
This depends greatly on where you are calling. Most telco's will lookup the name based on the number and will not use the name sent though the ss7 network. One way to test this is to call your self on you PRI. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain McBride Sent: Wednesday, September 10, 2008 12:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue Hi Folks, I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped. >From searching the list archives, I believe that a DMS wants to receive the CNAM info in the Q.931 call setup message. I've tried all permutations of switchtype (dms100 & national) and facilityenable that I can think of, but I still don't see CNAM coming out the other side. Telco confirms that "Name Out" is enabled on our PRI. Any pointers on what I'm missing, and/or how to debug further? zapata.conf: --- [channels] context=default callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no canpark=yes cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 ;facilityenable=yes immediate=no priindication=outofband switchtype=national context=from-pstn group=0 signalling=pri_cpe channel =>1-23 --- Regards, Iain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue
On Wed, Sep 10, 2008 at 01:21:42PM -0400, Iain McBride wrote: > I'm trying to send CallerID Name information out to the PSTN via a PRI > with Bell Canada with no success. With inbound calls (originating from > the PSTN) CNAM is received successfully, and we've not had any similar > problems with other Telco PRIs, so I'm stumped. > > >From searching the list archives, I believe that a DMS wants to receive > the CNAM info in the Q.931 call setup message. I've tried all > permutations of switchtype (dms100 & national) and facilityenable that I > can think of, but I still don't see CNAM coming out the other side. It was only from reading the other folo to your query that I found out that *any carrier* *ever* gets CNAM to send it to an FXS by any means *other* than a database dip, myself. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Write Asterisk CDR MySQL records to multiple servers
Hi to all, I actually have an asterisk server configured to write CDR mysql records in the same machine (localhost), but I want to write this records to another machine also in mysql at the same time, It is possible? It means that I want save the records in both machines. Thanks in advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
Ricardo Melendez wrote: > > Hi to all, I actually have an asterisk server configured to write CDR > mysql records in the same machine (localhost), but I want to write > this records to another machine also in mysql at the same time, It is > possible? It means that I want save the records in both machines. > > One way of doing is this is to setup the second machine as a MySQL slave for your current machine. You can specify which MySQL databases/tables you want to be mirrored, you can find lots of tutorials how to do this on the Web. HTH, Vahan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote: > Hi to all, I actually have an asterisk server configured to write CDR mysql > records in the same machine (localhost), but I want to write this records > to another machine also in mysql at the same time, It is possible? It > means that I want save the records in both machines. You can either use MySQL replication or you can use 2 different CDR drivers at the same time, such as ODBC, with the Mysql-ODBC-Connector and the MySQL CDR driver. Also, in 1.6, cdr_adaptive_odbc allows you to specify multiple CDR backends within the same configuration file. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multipleservers
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Tilghman Lesher > Sent: Wednesday, September 10, 2008 5:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Write Asterisk CDR MySQL > records to multipleservers > > On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote: > > Hi to all, I actually have an asterisk server configured to > write CDR mysql > > records in the same machine (localhost), but I want to > write this records > > to another machine also in mysql at the same time, It is > possible? It > > means that I want save the records in both machines. > > You can either use MySQL replication or you can use 2 > different CDR drivers at > the same time, such as ODBC, with the Mysql-ODBC-Connector > and the MySQL CDR > driver. Also, in 1.6, cdr_adaptive_odbc allows you to > specify multiple CDR > backends within the same configuration file. > > -- > Tilghman > It's also likely that you could use MySQL Proxy to achieve the result you want. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make outgoing call from Pri ?
Can you please show your CLI output for the following commands: zap show status pri show span x (where x is the number corresponding to the span configured as the PRI) as well as your relevant /etc/asterisk/extensions.conf So we can help you a bit more. On Wed, Sep 10, 2008 at 7:55 AM, Hiren Mistry <[EMAIL PROTECTED]> wrote: > Dear Everyone, > >I have a one Outgoing pri. And I have to use this pri for > outgoing call from this premises. Can anyone guide me how it is done and > what I have to mention in extention.conf If I have to get zero dialing > facility for out going. > > -- > With Regards, > Hiren Mistry > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to get the Call-ID into the CDR?
Here's the use case: call comes in, extension match is made on caller ID and dialed number, dial plan dials a number and connects the two call legs. Is there a way to get the Call-ID from the SIP header of the outbound call leg and store it in the CDR? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Resilience using DNS or phone feature ?
Oliver, We use DNS SRV records combined with short TTLs to provide failover. Thankfully, we have only used it when moving phones from one server to another in preparation for upgrades, but it worked like a champ then. CP On Wed, 2008-09-10 at 15:02 +0200, Olivier wrote: > Hi, > > I'm planning to deploy SIP hardphones in a serverless location. > Phones would be connected to 2 different Asterisk servers, one backing > up the other. > > I would like to offer resilience and I'm wondering about the best way > to do it. > > Phones themselves can register to a backup SIP proxy if first proxy > fails but, AFAIK, can't fall back to main server from backup server > when main server recovers. > > I'm wondering if should use DNS, phone multi-registration feature, or > a combination of DNS and phone multi-registration feature, to > implement resilience. > Your opinion ? > > Cheers > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zero time (nearly) call transfer.
If anyone would critique this method to achieve the following I'd be much obliged: THE EASY PROBLEM (solved): 1. I am the called party. 2. Call comes in and is answered by one of several endpoints 3. In the event that the call was answered by endpoint X (x being a local channel, dialed with the T option), the call can be transferred to an alternate endpoint by pressing the DTMF sequence as configured in features.conf Simple enough... THE HARDER PROBLEM (unsolved): Ideally, I'd like to accomplish the same thing but make the transfers instantaneous and near completely transparent to the calling party by doing something like this: DTMF sequence is pressed by called party, which triggers asterisk to establish a second channel to the alternate endpoint in advance. When the alternate endpoint answers, the calling party is 'switched' to the alternate endpoint, and the endpoint is dropped. In other words, a user wants to switch endpoints (usually wireless to terrestrial, or wireless-to-wireless) without troubling the customer--eliminating the "Hold on a sec while I transfer you: click--pause--ring--ring--hello?--OK--As I was saying..." I was thinking of a meetme conference and somehow triggering the drop of the first endpoint upon joining of the second endpoint. OR establishing the second channel connected to a park position, and then quickly transferring the call to that park position eliminating the call setup time. Could someone critique these methods? Can anyone think of a better way to do this? Is there an essential element here I'm missing that would make this impossible? I'm sure that with all the extremely clever people here, someone will have a good idea. I appreciate your help! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
*CLI> core show version Asterisk 1.4.13 built by root @ machine1 on a i686 running Linux on 2008-09-10 06:46:17 UTC Thanks Steve. What syntax should I use then? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Steve Murphy > Sent: Thursday, 11 September 2008 2:13 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint > > On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote: > > I am struggling to find out how to code hint in AEL2. > > > > I did hint(Custom:light1) and it keeps complaining about the : (colon). > > It works fine for SIP device like hint(SIP/439). > > > > Anyone who has tried it before? > > Yes, a while back I upgraded AEL to handle both ':' and '&' inside > the hint parens. This should work on 1.4 on up. What version of > asterisk are you using? 1.2? > > murf > > > -- > Steve Murphy > Software Developer > Digium ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote: > On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote: >> Hi to all, I actually have an asterisk server configured to write >> CDR mysql >> records in the same machine (localhost), but I want to write this >> records >> to another machine also in mysql at the same time, It is possible? >> It >> means that I want save the records in both machines. > > You can either use MySQL replication or you can use 2 different CDR > drivers at > the same time, such as ODBC, with the Mysql-ODBC-Connector and the > MySQL CDR > driver. Also, in 1.6, cdr_adaptive_odbc allows you to specify > multiple CDR > backends within the same configuration file. > > Are there any sample config's explaining how to setup cdr_adaptive_odbc? Is cdr_adaptive_odbc used with cdr.conf? cdr.conf makes no mention of how to use cdr_adaptive_odbc, yet cdr_adaptive_odbc.com doesn't explain how to set things like usegmtime or loguniqueid. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] distinguish trunk from same host
Hi All, I have multi-tenant setup, each customer have their own trunk, but goes to the same host [trunk-customer-1] host=10.10.10.213 type=peer context=from-trunk-customer-1 [trunk-customer-2] host=10.10.10.213 type=peer context=from-trunk-customer-2 I have no issues going outbound, but inbound has issues. [from-trunk-customer-1] exten => 888,1,Goto(local-ext|100100|1) [from-trunk-customer-1] exten => 888,1,Goto(local-ext|101100|1) my prob is when call from outside goes to 888 it's going in on the trunk-customer-1 instead of trunk-customer-2. Got these error message: "Call from 'trunk-customer-1' to extension '888' rejected because extension not found." hope someone can help. Thank you. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
On Wednesday 10 September 2008 19:55:15 Eric Chamberlain wrote: > On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote: > > On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote: > >> Hi to all, I actually have an asterisk server configured to write > >> CDR mysql > >> records in the same machine (localhost), but I want to write this > >> records > >> to another machine also in mysql at the same time, It is possible? > >> It > >> means that I want save the records in both machines. > > > > You can either use MySQL replication or you can use 2 different CDR > > drivers at > > the same time, such as ODBC, with the Mysql-ODBC-Connector and the > > MySQL CDR > > driver. Also, in 1.6, cdr_adaptive_odbc allows you to specify > > multiple CDR > > backends within the same configuration file. > > Are there any sample config's explaining how to setup > cdr_adaptive_odbc? Is cdr_adaptive_odbc used with cdr.conf? cdr.conf > makes no mention of how to use cdr_adaptive_odbc, yet > cdr_adaptive_odbc.com doesn't explain how to set things like usegmtime > or loguniqueid. I thought that the sample cdr_adaptive_odbc.conf was rather clear, but apparently not. The point of this module is to allow you log whatever you like in terms of the CDR variables. Do you want to log uniqueid? Then simply ensure that your table has that column. If you don't want the column, ensure that it does not exist in the table structure. If you'd like to call uniqueid something else in your table, simply provide an alias in the configuration file that maps the standard CDR field name (uniqueid) to whatever column name you like. Perhaps you'd like some extra CDR values logged that aren't in the standard repertoire of CDR variables (some that come to mind are certain values used for LCR: route, per_minute_cost, and per_minute_price). Simply set those CDR variables in your dialplan, i.e. Set(CDR(route)=27), ensure that a corresponding column exists in your table, and cdr_adaptive_odbc will do the rest. I do agree that I have overlooked gmtime as a possible setting for datetime fields in cdr_adaptive_odbc, and that's probably something that I need to add. However, I think that the method by which specifying which columns you'd like to have is certainly much more intuitive than the old "These are your columns. You must have them, or all CDRs will fail" approach. Hopefully, you will also see what I think is a rather innovative approach to CDRs and wonder how you ever got along without it. In fact, the adaptive approach has been now ported to most of the other CDR drivers, including mysql, postgres, and sqlite, and another developer (who is more familiar with that API) is working on TDS support. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
Ricardo Melendez wrote:
> Hi to all, I actually have an asterisk server configured to write CDR
> mysql records in the same machine (localhost), but I want to write
> this records to another machine also in mysql at the same time, It is
> possible? It means that I want save the records in both machines.
>
>
>
> Thanks in advance.
>
>
>
> Ricardo Melendez
>
I would customize my own cdr_addon_mysql.so by editing cdr_addon_mysql.c
and recompiling. It is fairly trivial to add a second MySQL server to
the mix. The query is on line 248 and it is:
if (mysql_real_query(&mysql, sqlcmd, strlen(sqlcmd))) {
Just define another MySQL connection and insert an addicional query.
Andres
http://www.neuroredes.com
>
>
>___
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>
>AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>Register Now: http://www.astricon.net
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Register Now: http://www.astricon.net
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